Files
vllm-omni/tests/entrypoints/openai_api/test_codec_stream_protocol.py
T
tzhouamandClaude Fable 5.1 49a839ecf9 [Edge][WP5] Add codec-token streaming mode with on-device decoding
* qwen3_tts_talker_only pipeline (single terminal LLM_AR stage,
  final_output_type "latent") + vllm_omni/deploy/qwen3_tts_talker_only.yaml;
  a terminal talker publishes its per-step codes.audio rows to the client
  (omni_client_multimodal_output_keys).
* /v1/audio/speech/stream: session.config.output_mode="codec_tokens" streams
  codec.start (JSON) + binary frames (<IHB seq/n_frames/flags + int16 rows,
  vllm_omni/entrypoints/openai/codec_stream.py) + codec.done; new input.cancel
  message aborts an in-flight generation for both codec and PCM sessions
  (single reader task + session inbox replaces direct socket reads); optional
  session.config.seed; audio.done gains "cancelled" only when it happened.
* serving_speech._generate_codec_chunks aligns the cumulative code tensor to
  output token ids (drops prefill placeholder/EOS rows) and emits deltas.
* TTS adapters: supports_codec_stream / codec_stream_spec (Qwen3-TTS: 16
  codebooks, 2048 ids, 12.5 Hz, 24 kHz, decoder qwen3_tts_tokenizer_v2_12hz).
* examples/online_serving/text_to_speech/qwen3_tts/edge_decoder_client.py
  decodes locally with Qwen3TTSTokenizerV2Decoder (ramp schedule, 72-frame
  context, cancel/reset), codec_parity_check.py and
  benchmarks/tts/codec_stream_e2e.py compare against the two-stage server.
* docs/serving/codec_token_streaming.md; CPU unit tests for the wire format,
  alignment, session config and handler (frames, unsupported adapter, cancel);
  L3 e2e tests/e2e/online_serving/test_qwen3_tts_codec_stream.py.

Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com>
2026-09-09 11:10:13 +08:00

184 lines
8.3 KiB
Python

"""Codec-token streaming: wire format, row alignment, and handler behaviour (CPU)."""
import numpy as np
import pytest
import torch
from fastapi import FastAPI, WebSocket
from vllm_omni.entrypoints.openai import codec_stream as cs
from vllm_omni.entrypoints.openai.protocol.audio import StreamingSpeechSessionConfig
pytestmark = [pytest.mark.core_model, pytest.mark.cpu]
def test_pack_unpack_roundtrip_and_flags():
codes = np.random.randint(0, 2048, size=(25, 16)).astype(np.int64)
buf = cs.pack_codec_frame(7, torch.from_numpy(codes), eos=False, segment_end=True)
assert len(buf) == cs.HEADER_SIZE + 25 * 16 * 2
seq, n, flags, arr = cs.unpack_codec_frame(buf, 16)
assert seq == 7 and n == 25 and flags == cs.FLAG_SEGMENT_END
assert arr.dtype == np.int16 and np.array_equal(arr.astype(np.int64), codes)
# EOS with zero frames
buf = cs.pack_codec_frame(8, None, eos=True, codebooks=16)
seq, n, flags, arr = cs.unpack_codec_frame(buf, 16)
assert (seq, n, flags) == (8, 0, cs.FLAG_EOS) and arr.shape == (0, 16)
buf = cs.pack_codec_frame(2**32 + 5, np.zeros((1, 16), dtype=np.int64), reset=True)
assert cs.unpack_codec_frame(buf, 16)[0] == 5
assert cs.unpack_codec_frame(buf, 16)[2] == cs.FLAG_RESET
def test_pack_rejects_bad_shapes_and_ranges():
with pytest.raises(ValueError):
cs.pack_codec_frame(0, np.zeros((2, 2, 2), dtype=np.int64))
with pytest.raises(ValueError):
cs.pack_codec_frame(0, np.full((1, 16), 40000, dtype=np.int64))
with pytest.raises(ValueError):
cs.unpack_codec_frame(b"\x00\x01", 16)
with pytest.raises(ValueError):
cs.unpack_codec_frame(cs.pack_codec_frame(0, np.zeros((1, 16), dtype=np.int64))[:-2], 16)
def test_align_codec_rows_drops_prefill_placeholder_and_eos():
# 1 prefill placeholder row (zeros, emitted before any token) + 3 real frames + EOS row
codes = torch.tensor([[0] * 16, [5] * 16, [6] * 16, [7] * 16, [0] * 16])
token_ids = [5, 6, 7, 2150] # codebook-0 ids then EOS
rows = cs.align_codec_rows(codes, token_ids, 2048)
assert rows.shape == (3, 16) and rows[:, 0].tolist() == [5, 6, 7]
# trailing -1 placeholders in token ids are ignored
rows = cs.align_codec_rows(codes, token_ids + [-1, -1], 2048)
assert rows.shape == (3, 16)
# out-of-range rows are dropped
bad = torch.tensor([[5] * 16, [3000] * 16])
assert cs.align_codec_rows(bad, [5, 6], 2048).shape == (1, 16)
assert cs.align_codec_rows(None, [], 2048).shape == (0, 0)
assert cs.align_codec_rows(torch.zeros(0, 16, dtype=torch.long), [], 2048).shape == (0, 16)
def test_session_config_codec_mode_validation():
cfg = StreamingSpeechSessionConfig(output_mode="codec_tokens", stream_audio=True, response_format="pcm")
assert cfg.output_mode == "codec_tokens"
with pytest.raises(ValueError):
StreamingSpeechSessionConfig(output_mode="codec_tokens", stream_audio=False, response_format="pcm")
with pytest.raises(ValueError):
StreamingSpeechSessionConfig(
output_mode="codec_tokens", stream_audio=True, response_format="pcm", word_timestamps=True
)
assert StreamingSpeechSessionConfig().output_mode == "pcm"
def test_codec_start_message_fields():
spec = {"codebooks": 16, "codebook_size": 2048, "frame_rate_hz": 12.5, "sample_rate": 24000, "decoder_id": "x"}
msg = cs.codec_start_message(utterance_index=1, sentence_index=0, sentence_text="hi", spec=spec, model="m")
assert msg["type"] == "codec.start" and msg["codebooks"] == 16 and msg["header_format"] == cs.HEADER_FMT
assert msg["decoder_id"] == "x" and msg["version"] == cs.CODEC_STREAM_VERSION
# --------------------------------------------------------------------- handler
def _codec_app(mocker, *, supports: bool = True, cancel_after: int | None = None):
from vllm_omni.entrypoints.openai.serving_speech import OmniOpenAIServingSpeech
from vllm_omni.entrypoints.openai.serving_speech_stream import OmniStreamingSpeechHandler
speech_service = mocker.MagicMock(spec=OmniOpenAIServingSpeech)
speech_service.forced_aligner_enabled = False
speech_service._prepare_speech_generation = mocker.AsyncMock(return_value=("req-1", object(), {}))
adapter = mocker.MagicMock()
adapter.supports_codec_stream = supports
adapter.codec_stream_spec = {
"codebooks": 16,
"codebook_size": 2048,
"frame_rate_hz": 12.5,
"sample_rate": 24000,
"decoder_id": "d",
}
speech_service._get_tts_adapter = mocker.MagicMock(return_value=adapter)
speech_service.engine_client = mocker.MagicMock()
speech_service.engine_client.abort = mocker.AsyncMock()
async def mock_codec_chunks(_generator, _request_id, *, tts_params=None, min_frames=1):
import asyncio
for i in range(3):
await asyncio.sleep(0.01)
yield np.full((1 + i, 16), i, dtype=np.int64), i == 2
speech_service._generate_codec_chunks = mock_codec_chunks
handler = OmniStreamingSpeechHandler(speech_service=speech_service, idle_timeout=5.0, config_timeout=5.0)
app = FastAPI()
@app.websocket("/v1/audio/speech/stream")
async def ws_endpoint(websocket: WebSocket):
# NOTE: no ``from __future__ import annotations`` in this module: FastAPI
# must resolve the WebSocket annotation, or it treats it as a query param.
await handler.handle_session(websocket)
return app, speech_service
def test_codec_stream_handler_frames_and_done(mocker):
from starlette.testclient import TestClient
app, svc = _codec_app(mocker)
with TestClient(app) as client, client.websocket_connect("/v1/audio/speech/stream") as ws:
ws.send_json(
{"type": "session.config", "output_mode": "codec_tokens", "stream_audio": True, "response_format": "pcm"}
)
ws.send_json({"type": "input.text", "text": "hello"})
ws.send_json({"type": "input.done"})
start = ws.receive_json()
assert start["type"] == "codec.start" and start["codebooks"] == 16
frames = []
while True:
msg = ws.receive()
if "bytes" in msg and msg["bytes"] is not None:
frames.append(cs.unpack_codec_frame(msg["bytes"], 16))
continue
done = __import__("json").loads(msg["text"])
break
assert [f[1] for f in frames] == [1, 2, 3]
assert [f[0] for f in frames] == [0, 1, 2]
assert frames[-1][2] & cs.FLAG_EOS
assert done["type"] == "codec.done" and done["total_frames"] == 6 and done["cancelled"] is False
assert ws.receive_json()["type"] == "session.done"
svc.engine_client.abort.assert_not_awaited()
def test_codec_stream_rejected_when_adapter_lacks_support(mocker):
from starlette.testclient import TestClient
app, _ = _codec_app(mocker, supports=False)
with TestClient(app) as client, client.websocket_connect("/v1/audio/speech/stream") as ws:
ws.send_json(
{"type": "session.config", "output_mode": "codec_tokens", "stream_audio": True, "response_format": "pcm"}
)
ws.send_json({"type": "input.text", "text": "hello"})
ws.send_json({"type": "input.done"})
err = ws.receive_json()
assert err["type"] == "error" and "not supported" in err["message"]
def test_input_cancel_mid_stream_aborts_request(mocker):
from starlette.testclient import TestClient
app, svc = _codec_app(mocker)
with TestClient(app) as client, client.websocket_connect("/v1/audio/speech/stream") as ws:
ws.send_json(
{"type": "session.config", "output_mode": "codec_tokens", "stream_audio": True, "response_format": "pcm"}
)
ws.send_json({"type": "input.text", "text": "hello"})
ws.send_json({"type": "input.done"})
assert ws.receive_json()["type"] == "codec.start"
ws.send_json({"type": "input.cancel"})
types = []
while True:
msg = ws.receive()
if "bytes" in msg and msg["bytes"] is not None:
continue
payload = __import__("json").loads(msg["text"])
types.append(payload["type"])
if payload["type"] == "session.done":
break
if payload["type"] == "codec.done":
assert payload["cancelled"] is True
assert "codec.done" in types and types[-1] == "session.done"
svc.engine_client.abort.assert_awaited()